There presently exists a significant amount of interest in the adaptation of packet switched network technologies, such as, for example, Frame Relay, IP and ATM, to carrying voice communication (e.g., telephone calls). Of the different packet switched network technologies, hereafter simply referred to as data network technologies, IP (Internet Protocol) is becoming increasingly wide spread and widely supported, mainly due to the growth of the Internet. Accordingly, the data networks which underlie the IP structure of the Internet have been growing at a much faster rate than the dedicated voice networks of the telephone companies. Soon the amount of data traffic will exceed that of voice traffic. As a result of this trend, more and more voice is being sent over data networks (e.g., Voice over IP) than data is being sent over voice networks (via V.34 and V.90 modems).
As use of the public Internet exploded in the mid-1990s, many users began implementing IP-based networks specifically adapted for voice over IP (VoIP) applications. To support such users, equipment manufacturers are developing products to enable inexpensive, universal voice over data networks.
Although significant progress has been made in the ad hoc engineering of data networks to carry voice as well as data, market requirements soon demanded a true convergence of these technologies into a single and ubiquitous communications service without being limited by the underlying technology. The widespread adoption of VoIP applications is dependent upon development of interconnection and interworking standards in order to deliver voice services ubiquitously over IP.
IP is based upon packet or cell switching technologies. This is in contrast to the public telephone network, which is a circuit switching technology, designed to carry voice transmissions. The packet switching and cell switching networks perform statistical multiplexing, wherein they dynamically allocate bandwidth to various links based on their transmission activity. Since bandwidth is not reserved for any specific path, the available bandwidth is allotted according to network needs at any particular time.
The traditional voice (or circuit switching) network uses a path dedicated to the transmission for the duration of the call, which is sent in a continuous bit stream. The line is monopolized by a call until it is terminated, even when the caller is put on hold and during periods of silence. Although this guarantees reliable and immediate transmission of voice, it results in very inefficient use of bandwidth. A line that is dedicated to the telephone cannot be utilized by other data even when there are no voice transmissions.
In contrast, data networks were originally designed to handle bursty data traffic, and as such, packet switching networks are inherently less efficient than the circuit switching network in dealing with voice. To achieve good voice quality, the delay of voice packets across the network must be minimal. Due to the shared nature of the packet/cell switching network, it might take time for transmissions to travel across the network. A transmission can be delayed because of network congestion. For example, it might “get stuck” behind a long data transmission that delays other packets. Network congestion can also result in dropped packets, which also detrimentally affects the integrity of voice transmissions.
Unfortunately, unlike most data applications, voice is very sensitive to delay. Good voice quality provides a faithful recreation of the conversation, with the same tone, inflection, pauses and intonation used by the speakers. Long and variable delays between packets result in unnatural speech and interfere with the conversation. Dropped packets result in clipped speech and poor voice quality. Fax transmissions are even more sensitive to the quality of the transmission and are less tolerant of dropped packets than voice. To improve the quality of voice transmission over data networks, quality of service standards are being developed.
Quality of service, or QoS, generally refers to the ability to define a level of performance in a data network. For example, some types of data networks specify modes of service that ensure optimum performance for latency sensitive traffic such as real-time voice and video. QoS has become a major issue on the Internet as well as in enterprise networks, because voice and video are increasingly traveling over IP-based data networks that were not designed for continuous speech or viewing.
RSVP (Reservation Protocol) is a well known IETF (Internet Engineering Task Force) communications protocol for implementing QoS between nodes within IP networks. RSVP primarily functions by signaling a router to reserve bandwidth for real-time transmission. RSVP is designed to clear a path for audio and video traffic eliminating annoying skips and hesitations. It has been sanctioned by the IETF, due to the fact that voice traffic is expected to increase dramatically on the Internet.
However, RSVP as currently defined imposes a significant penalty with regard to data network traffic overhead. A significant amount of data overhead is required for managing RSVP compliant QoS voice streams. For example, one prior art RSVP implementation uses a scheme where a unique RSVP session is created with an individual reservation for each voice stream. This is inefficient for bandwidth utilization because it generates separate “Path” and “RESV” packet streams (including periodic updates) for each voice stream. For example, as currently implemented, the “PSB” and “RSB” databases on each router along a voice stream path must store a unique record for each Path and Reservation respectively, and each Reservation produces a unique flow descriptor which adds to resource utilization. In a large network, this scheme leads to unmanageable overhead and resource utilization.
Thus, what is required is a method and system for efficiently implementing QoS for multiple voice streams over IP networks. What is required is a solution that reduces the amount of overhead traffic and resource utilization involved in implementing multiple voice streams over IP networks. Additionally, the required solution should significantly reduce RSVP signaling overhead for VoIP calls between VoIP dial-peers on IP network platforms. The present invention provides a novel solution to the above requirements.